Programmable auditory prosthesis with trainable automatic adaptation to acoustic conditions

ABSTRACT

An auditory prosthesis ( 30 ) comprising a microphone ( 27 ) for receiving the sound and producing a microphone signal responding to the received sound, an output device for providing audio signals in a form receivable by a user of the prosthesis ( 30 ), a sound processing unit ( 33 ) operable to receive the microphone signal and carry out a processing operation on the microphone signal to produce an output signal in a form suitable to operate the output device, wherein the sound processing unit ( 33 ) is operable in a first mode in which the processing operation comprises at least one variable processing factor which is adjustable by a user to a setting which causes the output signal of the sound processing unit ( 33 ) to be adjusted according to the preference of the user for the characteristics of the current acoustic environment.

This application is a continuation-in-part of PCT/AU03/00613, entitled“Programmable Auditory Prosthesis with Trainable Automatic Adaptation toAcoustic Conditions,” filed Nov. 27, 2003. The entire disclosure andcontents of the above application is hereby incorporated by reference.

BACKGROUND

1. Field of the Invention

The present invention relates to a programmable auditory prosthesis,such as a hearing aid or cochlear implant. In particular, the inventionis an auditory prosthesis that adjusts its sound processingcharacteristics in a particular acoustic environment in a manner that issimilar or identical to that previously determined by the user of theprosthesis as optimal for that environment.

2. Related Art

Hearing loss, which may be due to many different causes, is generally oftwo types, conductive and sensorineural. Of these types, conductivehearing loss occurs where the normal mechanical pathways for sound toreach the hair cells in the cochlea are impeded, for example, by damageto the ossicles. Conductive hearing loss may often be helped by use ofconventional hearing aid systems, which comprise a microphone, anamplifier and a receiver (miniature speaker) for amplifying detectedsounds so that acoustic information does reach the cochlea and the haircells. Since the elevation of the minimum detectable sound pressurelevel may vary with the frequency of an acoustic test stimulus, theamplifier may be preceded by, or comprise of, a bank of filters toenable different frequency components of the signal to be amplified bydifferent amounts.

Sensorineural hearing loss occurs where the hair cells in the cochleaand the attached auditory nerve fibres are damaged or destroyed.Sensorineural hearing loss results in an increase in the minimumdetectable sound pressure level, which often varies with the frequencyof the test stimulus. However, in contrast to conductive hearing loss,the sound pressure level that is uncomfortably loud at a given testfrequency is often approximately the same as for people with normalhearing. The result is a reduction in the dynamic range of soundpressure levels that are audible yet not uncomfortably loud with theimpaired ear, and this dynamic range of the impaired ear may varyconsiderably with the frequency of the acoustic test stimulus. For thisreason, sensorineural hearing loss is often treated with hearing aidsystems that employ non-linear amplification to compress the dynamicrange of common sounds towards the dynamic range of the impaired ear.Such systems may use a filter bank that is followed by a bank ofcompressive amplifiers, so that the dynamic range of the signal isreduced by an amount that is considered appropriate for the dynamicrange of the impaired ear in each band.

In many people who are profoundly deaf, the reason for deafness isabsence of, or destruction of, the hair cells in the cochlea whichtransduce acoustic signals into nerve impulses. These people are thusunable to derive suitable benefit from hearing aid systems, no matterhow much the acoustic stimulus is amplified, because there is damage toor absence of the mechanism for nerve impulses to be generated fromsound in the normal manner. It is for this purpose that cochlear implantsystems have been developed. Such systems bypass the hair cells in thecochlea and directly deliver electrical stimulation to the auditorynerve fibres, thereby allowing the brain to perceive a hearing sensationresembling the natural hearing sensation normally delivered to theauditory nerve. U.S. Pat. No. 4,532,930, the contents of which areincorporated herein by reference, provides a description of one type oftraditional cochlear implant system.

Cochlear implant systems have typically consisted of two essentialcomponents, an external component commonly referred to as a processorunit and an internal implanted component commonly referred to as astimulator/receiver unit. Traditionally, both of these components havecooperated together to provide the sound sensation to a user.

The external component has traditionally consisted of a microphone fordetecting sounds, such as speech and environmental sounds, a speechprocessor that converts the detected sounds, particularly speech, into acoded signal, a power source such as a battery, and an externaltransmitter coil.

The coded signal output by the speech processor is transmittedtranscutaneously to the implanted stimulator/receiver unit situatedwithin a recess of the temporal bone of the user. This transcutaneoustransmission occurs via the external transmitter coil which ispositioned to communicate with an implanted receiver coil provided withthe stimulator/receiver unit. This communication serves two essentialpurposes, firstly to transcutaneously transmit the coded sound signaland secondly to provide power to the implanted stimulator/receiver unit.Conventionally, this link has been in the form of an RF link, but othersuch links have been proposed and implemented with varying degrees ofsuccess.

The implanted stimulator/receiver unit traditionally includes a receivercoil that receives the coded signal and power from the externalprocessor component, and a stimulator that processes the coded signaland outputs a stimulation signal to an intracochlea electrode assemblywhich applies the electrical stimulation directly to the auditory nerveproducing a hearing sensation corresponding to the original detectedsound.

Different users of auditory prostheses require differing outputs fromtheir prosthesis to suit their individual requirements. This is the caseeven when individual users may clinically be regarded as havingidentical hearing loss profiles, are utilising identical prostheses, andwhen exposed to essentially identical acoustic environments. Because ofthis, sound processing schemes for hearing aids and cochlear implantstypically contain a number of parameters for which the values can beadjusted to suit the requirements of individual users. Examples of suchparameters include the sensitivity to incoming sounds and the variationof the frequency response. Typically, the parameter values are selectedeither by the prosthesis user in everyday situations (eg. thesensitivity (volume), frequency response), or by the clinician at thetime the prosthesis is fitted (eg. the baseline frequency response andthe rate at which the frequency response and sensitivity vary as theinput level varies).

In more recent times, there has been a trend to provide auditoryprostheses with an increasing number of adjustable parameters that canor must be adjusted to optimise performance. This increase has, however,highlighted the problem that there does not always exist a reliableprescriptive method for selecting the optimum values for the individualuser, particularly as some optimum values may vary among individuals whohave hearing loss profiles that may clinically be regarded as identical.It is accordingly often necessary for the clinician to make adjustmentsto the prosthesis based on the user's reported experiences away from theclinic, and the need to return to the clinic for these adjustments canbe time consuming and inefficient.

One example of a hearing aid that can receive the impressions of ahearing aid user and take these into consideration during operation isdescribed in U.S. Pat. No. 5,604,812. This document describes a hearingaid that has a memory that can store so-called “unsharp inputs”, orimpressions of the hearing aid wearer about prevailing ambientconditions and/or the volume of the output of the hearing aid,prescribable algorithms, hearing loss data and hearing aidcharacteristic data. A fuzzy logic module uses this data and controlsignals from an input signal analysis unit to calculate the outputsetting parameters for the hearing aid. The behaviour rules of the fuzzylogic module and/or the prescribable algorithms are calculated from thedata stored in the aid's memory by a neural network, with the neuralnetwork being typically implemented on a personal computer that isconnected to the aid. A problem with such an aid is the complexity ofthe processing scheme and an undesirably large drain on the aid's powersource, typically an on-board battery. Further to this, such an aidwould require a large amount of clinical time to correctly “train”, asthe user does not have direct control over what is optimal, and “unsharpinputs” are used in the training of the aid rather than precise anddirect inputs.

Another example of a hearing aid that can receive the preferences of ahearing aid user and take these into consideration is described in U.S.Pat. No. 6,035,050. Such an aid requires the user to identify theircurrent listening environment by selecting one of a limited number oflistening situations on a remote unit. Should the user find that theircurrent listening situation does not readily fall within those providedfor selection, the benefit of such a aid becomes greatly reduced.Further to this, some listening situations on the remote unit, such as“at work”, may consist of a range of different acoustic environments andhence may not be acoustically well-defined. Thus, a neural network maynot be able to reliably recognise such listening situations from ananalysis of the microphone signal that, due to practical considerationssuch as memory limitations and the desirability of fast recognition of achange in listening situation, must be limited to a period of seconds ora few minutes. Therefore, after training it is possible that the neuralnetwork may misclassify a listening situation, especially in situationsthat are not acoustically well-defined, which can result in theapplication of amplification parameters that are unsuitable for thecurrent listening situation. In such a device focus is placed onattempting to categorise specific acoustic environments, rather thanmeasure the parameters of the environment and use these parametersdirectly in the processing scheme, as is the case with the presentinvention. Another disadvantage of this device is that a major lifestylechange, such as a new workplace that is acoustically different to theprevious workplace, may require different amplification parameters andhence a return visit to the clinic.

Another example of a hearing aid that can receive the preferences of ahearing aid user and take these into consideration is described in U.S.Pat. No. 6,044,163. This document describes a hearing aid that issimilar to the hearing aid described in U.S. Pat. No. 6,035,050. A majordifference is that the neural network is not restricted to selection ofsets of amplification parameters that are stored in a memory, but maydirectly set the value of individual amplification parameters. Thedisadvantages of the hearing aid described in this patent are similar tothose of the hearing aid described in U.S. Pat. No. 6,035,050.

The present invention is adapted to providing users with an auditoryprosthesis that is adaptable and can adjust its output to the user whenthat user is exposed to varying ambient acoustic environments.

Further, the present invention is adapted to providing users with anauditory prosthesis that can be ‘trained’ by the individual user toadapt its output to the user's preference in listening conditionsencountered away from a traditional clinical situation, thereby reducingthe clinical time required to fit an optimise a prosthesis to theindividual user.

Any discussion of documents, acts, materials, devices, articles or thelike which has been included in the present specification is solely forthe purpose of providing a context for the present invention. It is notto be taken as an admission that any or all of these matters form partof the prior art base or were common general knowledge in the fieldrelevant to the present invention as it existed before the priority dateof each claim of this application.

SUMMARY

According to a first aspect, there is disclosed an auditory prosthesiscomprising:

a microphone for receiving sound and producing a microphone signalcorresponding to the received sound;

an output device for providing audio signals in a form receivable by auser of the prosthesis;

sound processing means operable to receive the microphone signal andcarry out processing operations on the microphone signal to produce anoutput signal in a form suitable to operate the output device, and

user control means;

wherein the processing means is operable in a first mode in which theprocessing operation comprises at least one variable processing factorwhich is adjustable by a user using the user control means to acorresponding setting which causes the output signal of the soundprocessing means to be adjusted according to the preference of the userin the current acoustic environment; and

wherein the sound processing means is simultaneously operable in asecond mode in which at least one of the at least one variableprocessing factor(s) is automatically adjusted on the basis of one ormore previously selected settings.

Preferably, the control means can be adjusted throughout a substantiallycontinuous range of settings, with the preferred setting being used tocalculate the one or more variable processing factors, which alter theprocessing operations applied to the microphone signal.

Preferably, the settings of the control means are related to one or morevariable processing factors by defined mathematical relationshipsincorporating coefficients, and where the values of said coefficientsare calculated to be those values that cause the variable processingfactors to best approximate the variable processing factors thatoccurred as a result of previous settings of the control means.

In this invention, it will be appreciated that what is considered anoptimal output signal by one user may not be considered an optimalsignal by another user. The present invention ensures that theprosthesis operates in a manner that suits the requirements of the userfor the acoustic environments that the user encounters rather than insome manner predetermined from clinical tests which may not replicatethe type of acoustic environments to which the user is routinely oroccasionally exposed, and which may not necessarily replicate thesubjective reactions of users when in those environments.

Preferably, the data memory means stores data sets representative of thecharacteristics of at least one acoustic environment. Preferably the atleast one variable processing factor is automatically adjusted on thebasis of the acoustic environments in which at least one variableprocessing factor was selected.

In a first embodiment, the processing means may be adapted when operablein the first mode to offer two or more possible optimal settings of thevariable processing factor for selection by the user. In this case, theuser is preferably able to compare the operation of the prosthesis whenoperating in a particular acoustic environment when operating with eachsetting and then select from these the setting that is best for thatparticular environment. The setting of the variable processing factorthat is selected by the user may then be stored in the data memorymeans, optionally with data indicative of the characteristics of theparticular acoustic environment. This process may be repeatable so as toallow the processing means to monitor whether the user's preference fora particular setting of the variable processing factor changes with timeor usage. In this way, the user effectively selects or votes for thebest setting each time. By repeating the process, the number ofselections or votes made by the user for each setting may be monitoredto allow the prosthesis to select the optimal setting of the variableprocessing factor or optionally, the optimal setting of the variableprocessing factor for a particular environment.

In this embodiment, the user may alternate between listening with theoffered settings by operating a control means, and can select or votefor a setting of the variable processing factor(s) by operating anindicator means. In one embodiment, the control means may comprise aswitch or set of buttons that is actuable by the user, and the indicatormeans may comprise a switch or button that is actuable by the user.

In another embodiment, the processing operation of the processing meansmay be adjustable by the user when the processing means is in the firstmode of operation. Rather than offering a discrete selection of possibleoptimal settings for selection by the user, in this embodiment, the useris able to adjust a control means throughout a substantially continuousrange that allows the user to alter the variable processing factor(s)throughout a range of values, and thus alter the processing operation ofthe processing means. Once the user has adjusted the control means to asetting in the substantially continuous range that is considered by theuser an optimal setting for a particular acoustic environment, the usermay operate an indicator means, such as a switch or button, leading tostorage of that setting and optionally, data indicative of theparticular acoustic environment in the data memory means. Actuation ofthis indicator means may be taken by the processing means as indicatingthat the particular setting of the control means at that time isconsidered optimal by the user for the particular acoustic environmentin which the user finds themself.

In this embodiment, the control means may comprise a rotary wheelcontrol. The rotary wheel control may be mounted on the housing of theprosthesis or alternatively, may be mounted on a remote unit. Thecontrol means could also be in the form of a toggle switch or pressbuttons.

In the above embodiments, the indicator means may be mounted on ahousing of the prosthesis. In another embodiment, the indicator meansmay be mounted on a remote unit.

In a preferred embodiment, the settings selected by the user as beingoptimal to that user for a plurality of acoustic environments may bestored in the data memory means and optionally, with data indicative ofthe characteristics of those particular acoustic environments. Theprocessing means operates in the second mode, and may simultaneously beoperable in the first mode for a defined period of time or can beconsidered as operating in the first mode every time the user adjuststhe control means and selects what is considered a new optimal settingfor that particular environment. In another embodiment, the prosthesismay have a defined training period in which the prosthesis may beoperable in the first mode of operation. This period may be as long asthe user wishes or optionally, the prosthesis may signal to the userwhen the training period is complete. In this case, the prosthesis maybe adapted to indicate the training period is complete once there hasbeen a particular pre-defined number of instances in which the user hasnot selected a setting which is substantially different to the settingalready estimated by the prosthesis to be optimal.

In a preferred embodiment, the prosthesis may further comprise a soundanalysis means. The sound analysis means preferably receives inputsignals from the microphone and monitors the acoustic environment of theprosthesis user. The sound analysis means preferably provides an outputrepresentative of the acoustic environment being monitored at that time.

In a further embodiment, the data memory means may comprise one or moredata memory locations. In one embodiment, the data memory means maycomprise five data memory locations. In this embodiment, the first datamemory location may contain the audiometric data of the prosthesis userand/or individual data for one or more loudness models used by theprosthesis. The second data memory location may contain characteristicdata about the hearing prosthesis. The third data memory location maycomprise one or more equations used to predict the optimal processingoperation of the processing means for an individual user in differentacoustic environments. The fourth data memory location may store theoptimal sound processing data as selected by the user of the prosthesis.This data may be ordered in different ways, such as sequentially orindexed according to the corresponding acoustic environment datasupplied by the sound analysis means. The acoustic environment data thatcorresponds to the optimal sound processing data is optionally stored inthe fifth data memory location. Other data memory means having differingnumbers of memory locations may be envisaged.

In one embodiment, the fourth data memory location may store apredefined maximum number of sets of optimal sound processing data. Inone embodiment, the memory location may store a maximum of 400 datasets. Other maximum numbers can, however, be envisaged. In oneembodiment, the data processing means does not utilise all stored datasets but only a predefined number of most recently logged data sets. Forexample, the data processing means may only utilise the last 256 datasets when determining the optimal value of a variable processing factor.In one embodiment, the newest data set to be stored in the memorylocation can be stored instead of the oldest data set in the memorylocation. This first in first out storage system ensures only the mostrecently logged data is ever stored in the prosthesis at any one time.In another embodiment, older data cannot be overwritten such that oncethe memory location is full no further data sets can be logged by theprosthesis. In another embodiment, old data is preferably selectivelyoverwritten by the newest sound processing data according to theacoustic environment in which the sound processing data was selected.

In a further embodiment, the prosthesis preferably further comprises adata processing means. The data processing means preferably receives theoutput of the sound analysis means. Based on the output of the soundanalysis means, the data processing means can be adapted to calculatethe loudness of sounds present at the microphone. In one embodiment, thedata processing means can calculate the loudness of the sounds as theywould appear to a person with normal hearing and/or to a person with adegree of hearing impairment. The data processing means can be adaptedto calculate other acoustic and psychoacoustic measures of the soundspresent at the microphone. The data processing means can use theacoustic and/or psychoacoustic measures as inputs to the one or moreequations stored in the third data memory location which estimate theoptimal sound processing operation of the sound processing means for theuser in the acoustic environments represented by the sound analysismeans.

The data processing means further preferably uses the hearing prosthesischaracteristic data stored in the second data memory location and theoptimal sound processor data generated by the equations to automaticallyand continuously determine the appropriate setting of the processingmeans to provide the optimal output signal for the user in the currentacoustic environment being experienced by the user.

In a preferred embodiment, the processing means can include an amplifiermeans and a gain control means. In this case, the operation of theamplifier means is preferably adjustable when in the first mode to allowthe user to optimise the gain at each or every frequency of theamplifier in a particular acoustic environment. Once optimised, theprosthesis operates in a second mode in which the amplificationcharacteristics of the amplifier means automatically adjusts to theoptimal level. That is, in such embodiments the variable processingfactor is preferably the amplifier gain at each or every frequency.

In a preferred embodiment of the above aspects, the gain at each orevery frequency of the amplifier means is calculated through use of anequation having a pre-defined form. In one form the equation can be:

G _(i) =a _(i) +b _(i)*max(L _(i) ,c _(i))+d_(i)*(SNR_(i)−SNR_(av))  (1)

where

-   -   i=the frequency band number;    -   G_(i)=gain for band i;    -   a_(i)=trainable coefficient for band i;    -   b_(i)=trainable coefficient for band i;    -   c_(i)=trainable coefficient for band i;    -   d_(i)=trainable coefficient for band i;    -   L_(i)=sound pressure level at the microphone in band i;    -   SNR_(i)=signal to noise ratio in band i; and    -   SNR_(av)=average SNR in all bands.

Other forms of the equation with other acoustic or psychoacousticparameters, such as higher-order moments of the spectrum of the signaland variations of these moments with time, or other statisticalparameters or estimates of the acoustic signal or combinations of theseparameters, and optionally with other additional coefficients, cancalculate the gain or processing factors other than gain, such as thespeed at which the prosthesis reacts to change in the acousticenvironment, the choice of the equations used will depend upon theparticular application desired, with such applications falling withinthe scope of the present invention.

In the above embodiment, trainable coefficients include a, b, c and d,which as a result leads to the amplifier gain G being a variableprocessing factor.

Thus, in embodiments of the invention, more than one variable processingfactor of the processing means can be adjustable by the user using thecontrol means or multiple control means. Similarly, more than onevariable processing factor of the processing means can be automaticallyadjusted as appropriate to any acoustic environment. In this embodiment,the control means or multiple control means can allow the user to adjustone or more of the following example operations of the processor means:

(i) the volume of the output signal;

(ii) the gain of the output signal at particular frequencies relative toother frequencies, for example the mid frequency gain can be boosted orattenuated with respect to the low or high band frequencies of theoutput signal; and

(iii) a slope control where the low and high frequency band gains areadjusted in opposing directions while the mid band gain is unchanged.

In this embodiment, for each of the operations of the processor meansthe user can select a setting which is optimal for each variableprocessing factor for the particular acoustic environment that they arein by actuating the indicator means. On multiple operations of thecontrol means the user may select a setting which is optimal for one ormore variable processing factors for the acoustic environment that theuser experiences by actuating the indicator means. Thus, more than oneoperation of the sound processor means, as well as more than onevariable processing factor, can be adjusted in combination for each voteor selection. Each time the indicator means is actuated, the gain ineach frequency band is preferably recorded along with a data setindicative of the acoustic environment detected by the microphone. Thisdata can be used with previous sets of data by the data processing meansto calculate the gain equation coefficients that best match the loggedtraining data in each band. The recorded gain in each frequency band maybe used to calculate the gain equation co-efficients.

In a further embodiment, when operating in the first mode the dataprocessing means preferably does not calculate optimal gain equationcoefficients until a predetermined number of selections have been madeby the user. In one embodiment, the prosthesis can be adapted to notcalculate optimal gain equation coefficients until at least fiftyselections have been made by the user. In this case, where thepredetermined number of selections have not been made, the processingmeans may when operating in the second mode preferably automaticallycalculate the output appropriate to the current environment, based onpredetermined coefficients and/or variable processing factors. Once thepredetermined number of selections have been made, the processing meansmay calculate the optimal gain equation coefficients on the basis ofprevious user preferences when in the first mode of operation, andautomatically and continuously calculate the variable processing factorswith these equations when in the second mode of operation. In anotherembodiment, the data processing means preferably does not wait for apredetermined number of selections to be made by the user beforecalculating the optimal gain equation coefficients, but preferably doesnot use the calculated coefficients when in the second mode of operationuntil the variable processing factors that are preferred by the user canbe predicted with the equations and calculated coefficients with acertain degree of accuracy.

In an alternative embodiment, equations may not be used to determine theoptimal variable processing factors for different environments. Theoptimally adjusted variable processing factors may be stored in thefourth data memory at locations determined by the corresponding acousticenvironment data supplied by the sound analysis means. Prior to thecommencement of operation in the first mode, the memory locations arepreferably loaded with processing factors that are derived fromempirical observations or prescriptive procedures for the correspondingacoustic environment parameter values. When operating in the first modethe optimally adjusted processing factor(s) may be stored in the fourthdata memory at locations determined by the acoustic environment data.The optimally adjusted processing data may simply overwrite theprescribed data, or may be combined with old data written to the samelocation using a mathematical and/or statistical procedure. Whenoperating in the second mode, the data supplied by the sound analysismeans may be used to index the fourth data memory location, and thevalue(s) stored at the indexed memory location are preferably used toset the target value(s) of the corresponding processing factor(s) forthe sound processing means. The processing factor(s) applied in thesound processing means may slowly converge towards the target value(s)to avoid the possibility of undesirable acoustic effects associated withexcessively rapid or instantaneous changes of the applied value(s). Thisalternative embodiment eliminates the need for the fifth data memorylocation and the equations stored in the third data memory location, andhence the variation of a processing factor with the acoustic environmentis not restricted by the relationships that are predefined in theequations.

In another alternative embodiment, the optimally adjusted variableprocessing factor(s) may be stored in the fourth data memory locationand a mathematical and/or statistical procedure is used to calculate theoptimal value of the processing factor(s). This alternative embodimenteliminates the need for the equations stored in the third data memorylocation, and the storage of acoustic environment data either directlyin the fifth data memory location or to index setting data in the fourthdata memory location. This alternative embodiment is an appropriatesimplification for processing factors that do not vary with the acousticenvironment, such as the maximum permissible level of the output of ahearing aid or cochlear implant before the output is uncomfortably loud.

In a further embodiment, the user can train the prosthesis to provideoptimal variable processing factors for different listening criteria,such as to maximise speech intelligibility, listening comfort, orpleasantness. In one embodiment, an equation similar in form to equation1 can comprise acoustic environment or psychoacoustic parameters thatare highly correlated to a particular listening criterion, such asmaximising speech intelligibility or listening comfort. When operable inthe second mode the variable processing factors may automatically adaptfor the predicted weighted combination of listening criteria asdetermined by the values of the coefficients and the acousticenvironment and/or psychoacoustic parameters used by the equations.

In another embodiment, while operable in the first mode the user mayindicate the listening criterion or combination of criteria used todetermine the preferred adjustment of the variable processing factor(s)with an indicator means, or optionally the aid automatically predictsthe listening criterion or combination of criteria from the values ofacoustic environment and/or psychoacoustic parameters that are known tocorrelate with different listening criteria, and/or from the presence ofsignals of interest such as speech and music as detected by theprosthesis. The fourth data memory location (otherwise referred to as afirst data memory location in claim 76) may be divided into sections fordifferent listening criteria, and the optimally adjusted variableprocessing factor(s) may be stored in a memory location(s) as determinedby the listening criterion or combination of listening criteria underwhich they were adjusted by the user. The fifth data memory location(otherwise referred to as a further data memory location in claim 77) isoptionally also divided into sections for each different listeningcriterion for the storage of the acoustic or psychoacoustic parametervalues that correspond to the values stored in the fourth data memorylocation. With this structure, the data in each section of the fourthand fifth data memory locations is preferably used to calculate theoptimal coefficients of variable processing factor equations similar inform to equation 1 for each listening criterion. When operable in thesecond mode, the user can indicate the current listening criterion orcombination or criteria via an indicator means or alternatively, thelistening criterion or weighted combination of criteria can be predictedfrom the values of acoustic environment and/or psychoacoustic parametersthat are known to correlate with different listening criteria and/or thepresence of signals of interest such as speech and/or music, or by oneor more trainable listening criteria equations that have acousticenvironment or psychoacoustic parameters as their input, where theoptimal coefficients of each listening criterion equation are calculatedfrom the data stored in the corresponding sections of the fourth andfifth data memory locations. Once the listening criterion or weightedcombination of listening criteria are either indicated by the user orautomatically predicted by the prosthesis, the optimal variableprocessing factors may be calculated with equations similar in form toequation 1 for each criterion, and are combined according to theindicated or predicted weighting of the listening criteria.

In an alternative embodiment, when operable in the first mode thelistening criterion or combination of criteria indicated by the user viaan indicator means or predicted by the prosthesis may be used inaddition to the corresponding acoustic environment and/or psychoacousticparameters to index the location in the fourth data memory where theoptimally adjusted variable processing factor(s) are stored. Whenoperable in the second mode, the current values of acoustic environmentand/or psychoacoustic parameters and the indicated and/or predictedlistening criterion or combination or criteria may be used to index thefourth data memory location to retrieve the optimal variable processingfactor(s) for the prevailing conditions. In another embodiment, thelistening criterion or combination of criteria may not be used to indexthe fourth data memory location, and the processing factor value(s)stored at each location in the fourth data memory when operable in thefirst mode may be modified according to the weighted combination oflistening criteria for the particular acoustic environment.

In the above embodiments, the listening criterion or combination ofcriteria is indicated by the user by an indicator means, such as aswitch, toggle switch, or set of pushbuttons on the housing of theprosthesis, or alternatively on a remote unit.

According to a second aspect, the present invention provides an auditoryprosthesis comprising:

a microphone which receives sound and produces an output signalcorresponding to the received sound;

an output device that provides audio signals in a form receivable by auser of the prosthesis;

a sound processing means operable to receive the microphone signal andcarry out a processing operation on the microphone signal to produce anoutput signal in a form suitable to operate the output device, whereinthe processing means is operable in a first mode in which the processingoperation comprises at least one variable processing factor which isadjustable by a user to a setting which causes the output signal of theprocessing means to be adjusted according to the preference of the userfor the characteristics of the current acoustic environment;

a sound analysis means that also receives the microphone signal andoutputs a data set representative of the acoustic environment;

a data memory means for storing a plurality of said user adjustedsettings; and

a data processing means that analyses said stored settings and isoperable to output control signals to the sound processing means;

wherein, upon receiving control signals from the data processing means,the sound processing means is simultaneously operable in a second modein which said at least one variable processing factor is automaticallyadjusted on the basis of the previously adjusted variable processingfactor selected by the user.

In this aspect, the features of the prosthesis can have the features asdescribed above with reference to the first aspect where such featuresare compatible with the operation of the prosthesis as describedtherein.

In this aspect, the hearing prosthesis preferably uses the data memorymeans to store a record of optimal settings as determined by the user ofthe prosthesis for different acoustic environments. This record is thenused by the data processing means to calculate coefficients of one ormore equations that are used to predict the optimal setting of theprocessing operation of the speech processor for that user when exposedto a different acoustic environment.

In one embodiment, the hearing prosthesis can be adapted to output audiosignals to a single ear of the user. In another embodiment, the hearingprosthesis can be adapted to output audio signals to both ears of theuser. In yet a further embodiment, signals from the inputs of twoprosthesis can be connected to the sound analysis means by wire orwirelessly. The control means for the above embodiments can be locatedon the housing of each prosthesis, a remote unit for each prosthesis, orthe control means for both prostheses can be located on a common remoteunit.

In one embodiment of the above aspects, the hearing prosthesis can be ahearing aid. In this embodiment, the output device of the hearing aidcan be an earphone that receives the output signal of the processingmeans and generates amplified sounds that are delivered into the ear ofthe user.

In another embodiment, the hearing prosthesis can be an implantedhearing aid device whereby the output device of the prosthesis can be avibrating mechanism, mechanically coupled to the middle or inner ear.

In another embodiment of the above aspects, the hearing prosthesis canbe a cochlear implant. In this embodiment, the output device cancomprise a receiver/stimulator unit that receives encoded stimulationdata from the processing means and outputs stimulation signals that aredelivered to the cochlea through an electrode array. In this embodiment,the sound processing means can comprise a speech processor that uses acoding strategy to extract speech from the sounds detected by themicrophone before or after processing of the microphone signal accordingto the variable processing factors. In one embodiment, the speechprocessor of the cochlear implant can perform an audio spectral analysisof the acoustic signals and output channel amplitude levels. Thetransformation from the spectrum of the acoustic signal to electricalchannel amplitude levels may be determined by a variable processingfactor based on previously selected variable processing factors. Thethreshold and discomfort channel amplitude levels can also be processingfactors that are based on previous adjustments and selections made bythe prosthesis user. The sound processor can also sort the outputs inorder of magnitude, or flag the spectral maxima as used in the SPEAKstrategy developed by Cochlear Ltd.

The receiver/stimulator unit is preferably positioned within a housingthat is implantable within the user. The housing for thereceiver/stimulator unit is preferably implantable within a recess inthe bone behind the ear posterior to the mastoid.

The speech processing means is, in use, preferably mounted external tothe body of the user such that the signals are transmittedtranscutaneously through the skin of the user. Signals can preferablytravel from the processing means to the receiver means and vice versa.The receiver/stimulator unit can include a receiver coil adapted toreceive radio frequency (RF) signals from a corresponding transmittercoil worn externally of the body. The radio frequency signals cancomprise frequency modulated (FM) signals. While described as a receivercoil, the receiver coil can preferably transmit signals to thetransmitter coil which receives the signals.

The transmitter coil is preferably held in position adjacent theimplanted location of the receiver coil by way of respective attractivemagnets mounted centrally in, or at some other position relative to, thecoils.

During use, the microphone is preferably worn on the pinna of the user,however, other suitable locations can be envisaged, such as a lapel ofthe implantee's clothing. The speech processor preferably encodes thesound detected by the microphone into a sequence of electrical stimulifollowing given algorithms. The encoded sequence is transferred to theimplanted receiver/stimulator unit using the transmitter and receivercoils. The implanted receiver/stimulator unit may demodulate the FMsignals and allocates the electrical pulses to the appropriate attachedelectrode by an algorithm which is consistent with the chosen speechcoding strategy.

The external component housing the processing means further preferablyhouses a power supply. The power supply can comprise one or morerechargeable batteries. The transmitter and receiver coils may be usedto provide power via transcutaneous induction to the implantedreceiver/stimulator unit and the electrode array.

While the implant system can rely on external componentry, in anotherembodiment, the microphone, speech processor and power supply can alsobe implantable. In this embodiment, the implanted components can becontained within a hermetically sealed housing or the housing used forthe stimulator unit.

According to a third aspect there is provided a method of adjusting anauditory prosthesis by a user in an acoustic environment comprising thesteps of:

producing a microphone signal from a microphone that corresponds tosound received by the microphone;

providing audio signals, through an output device, in a form receivableby the user;

carrying out a processing operation, through a sound processing means,on the microphone signal so as to produce an output signal in a formsuitable to operate the output device wherein the processing means isoperable in a first mode in which the processing operation comprises atleast one variable processing factor which is adjustable by a user to asetting which causes the output signal of the sound processing means tobe adjusted according to the preference of the user for thecharacteristics of the current acoustic environment;

storing the setting in a data memory means;

wherein the sound processing means is simultaneously operable in asecond mode in which said at least one variable processing factor isautomatically adjusted on the basis of the previously adjusted variableprocessing factor selected by the user.

According to a fourth aspect there is provided a method of adjusting anauditory prosthesis by a user in an acoustic environment comprising thesteps of:

producing a microphone signal from a microphone that corresponds tosound received by the microphone;

providing audio signals, through an output device, in a form receivableby the user;

carrying out a processing operation, through a sound processing means,on the microphone signal so as to produce an output signal in a formsuitable to operate the output device wherein the processing means isoperable in a first mode in which the processing operation comprises atleast one variable processing factor which is adjustable by a user to asetting which causes the output signal of the sound processing means tobe adjusted according to the preference of the user for thecharacteristics of the current acoustic environment;

receiving the microphone signal at a sound analysis means thatsubsequently outputs a data set representative of the acousticenvironment of the prosthesis;

storing a plurality of user adjusted settings in a data memory means;

analysing the stored user adjusted settings and transmitting controlsignals to the sound processing means;

wherein, upon receiving the control signals, the sound processing meansis simultaneously operable in a second mode in which said at least onevariable processing factor is automatically adjusted on the basis of thepreviously adjusted variable processing factor selected by the user.

The present invention enables the user of an auditory prosthesis toadjust the operation of the processing means in a simple way while theyare using the prosthesis in a normal way. The invention furthereventually allows the user to ‘train’ the prosthesis to adjust itsoperation automatically. This has a number of advantages including:

(1) greater levels of user satisfaction for both hearing aids andcochlear implants, because of the better control users have over theirhearing;

(2) improved listening comfort, and/or speech intelligibility and/orsubjective sound quality, because processing parameters are optimised byeach user to the personal preference and needs of each user;

(3) the ability to generalise the ‘training’ algorithm to workeffectively with future devices that have different, or additional,parameters that require adjustment;

(4) because the training is carried out by the user in daily life,expensive clinical time is not required to achieve adjustment of thehearing aid and as adjustment is carried out in real life conditions,the adjustment better relates to actual performance of the device,rather than ideal clinical conditions; and

(5) The ability to generalise the training algorithm to work effectivelyin environments for which no training has been provided by the user, byautomatically applying the general relationships between preferredvariable processing factors and acoustic characteristics that have beenestablished by the training

BRIEF DESCRIPTION OF THE DRAWINGS

By way of example only, a preferred embodiment of the invention is nowdescribed with reference to the accompanying drawings, in which:

FIG. 1 is a pictorial representation of a prior art cochlear implantsystem;

FIG. 2 is a block diagram of one embodiment of an auditory prosthesisaccording to the present invention;

FIG. 3 is a flowchart setting one particular mode of operation of theprosthesis of FIG. 2;

FIG. 4 is a graph depicting results of experiments performed on aprototype prosthesis according to one embodiment of the presentinvention;

FIG. 5 is a diagram representing the storage of processing factor valuesin a two-dimensional memory space; and

FIG. 6 is a graph showing a relationship between input and output soundpressure levels at various stages of the training process according toone embodiment of the present invention.

DETAILED DESCRIPTION

While the present invention is not directed solely to a cochlearimplant, it is appropriate to briefly describe the construction of onetype of known cochlear implant system with reference to FIG. 1.

Known cochlear implants typically consist of two main components, anexternal component including a speech processor 29, and an internalcomponent including an implanted receiver and stimulator unit 22. Theexternal component includes a microphone 27. The speech processor 29 is,in this illustration, constructed and arranged so that it can fit behindthe outer ear 11. Alternative versions may be worn on the body. Attachedto the speech processor 29 is a transmitter coil 24 that transmitselectrical signals to the implanted unit 22 via a radio frequency (RF)link.

The implanted component includes a receiver coil 23 for receiving powerand data from the transmitter coil 24. A cable 21 extends from theimplanted receiver and stimulator unit 22 to the cochlea 12 andterminates in an electrode array 20. The signals thus received areapplied by the array 20 to the basilar membrane 8 and the nerve cellswithin the cochlea 12 thereby stimulating the auditory nerve 9. Theoperation of such a device is described, for example, in U.S. Pat. No.4,532,930.

As depicted diagrammatically in FIG. 1, the cochlear implant electrodearray 20 has traditionally been inserted into the initial portion of thescala tympani of the cochlea 12 up to about a full turn within thecochlea.

A block diagram depicting one embodiment of an auditory prosthesisaccording to the present invention is depicted generally as 30 in FIG.2. The block diagram depicts features of both a cochlear implant and ahearing aid. Despite this, it will be appreciated that the diagram isrepresentative only and that a prosthesis according to the presentinvention may not necessarily act as both a cochlear implant and hearingaid.

The auditory prosthesis 30 comprises a microphone 27 which receivessound and produces a microphone signal corresponding to the receivedsound and an output device that provides audio signals in a formreceivable by a user of the prosthesis 30. As can be seen, the outputdevice can comprise an earphone 31 in the case where the prosthesis 30is a hearing aid. Where the prosthesis 30 is a cochlear implant, theoutput device comprises an encoder/transmitter unit 32 that outputsencoded data signals to the external transmitter coil 24.

The prosthesis 30 further has a sound processing unit 33 that isoperable to receive the microphone signal provided by the microphone 27and produce an output signal in a form suitable to operate either theearphone 31 or the encoder/transmitter unit 32.

The nature and function of the sound processor 33 will depend on whetherthe prosthesis 30 is a cochlear implant or a hearing aid. In the case ofa hearing aid, the sound processor 33 includes at least an amplifierwhereas in the case of cochlear implant the sound processor 33 willinclude an amplifier and a speech processor that uses a coding strategyto extract speech from the sounds detected by the microphone 27. In thedepicted embodiment, the speech processor of the cochlear implant canperform an audio spectral analysis of the acoustic signals and outputchannel amplitude levels. The sound processor can also sort the outputsin order of magnitude, or flag the spectral maxima as used in the SPEAKstrategy developed by Cochlear Ltd. Other coding strategies could beemployed.

In this invention, the sound processor 33 comprises at least onevariable processing factor, such as amplification, which in a first modeof operation is adjustable to a setting in which the output signal ofthe sound processor 33 is optimised for the preference of the user in atleast one acoustic environment. For example, the amplificationcharacteristics of the amplifier can be adjusted by a user of theprosthesis 30 to suit the surrounding background noise in a particularenvironment.

The prosthesis 30 further comprises a data memory means 34 for storingat least data indicative of the setting selected by the user andoptionally data sets representative of the characteristics of theacoustic environment at the time the setting adjustment and selection ismade.

The sound processor 33 when operable in a first mode recalculates thetrainable coefficients associated with a corresponding variableprocessing factor after each adjustment by the user.

The sound processor 33 can be adapted to offer two or more possibleoptimal settings for selection by the user. One of these settings may bethe one automatically calculated by the processor and the other may be asystematic departure therefrom. In this case, the user is preferablyable to compare the operation of the prosthesis when operating in aparticular acoustic environment when operating with each setting andthen select from these the setting that is best for that particularenvironment. The setting that is selected by the user can then be storedin the data memory means 34 optionally with data indicative of theparticular acoustic environment. This process can be repeatable so as toallow the data processing unit 38 to gather the preferences and tomonitor whether the user's preference for a particular environmentchanges with time or usage. In this way, the user effectively selects orvotes for the best setting each time.

In this embodiment, the user can alternate between settings by operatinga control means 36 which can be a toggle switch or a set of pushbuttonsfor example, and select a setting by operating an indicator means whichis represented by switch 35 in FIG. 2. The switch 35 and control means36 can be mounted on the housing of the prosthesis 30, or while notdepicted, the switch and control means could be mounted on a remoteunit.

The processing operation of the sound processor 33 can be operable inanother manner or mode, in which its operation is adjustable by theuser. Rather than offering a selection of possible optimal settings forselection by the user, in this arrangement, the user is able to adjust acontrol means 36 that allows the user to alter the processing operationof the sound processor unit 33. Once the user has adjusted the controlmeans 36 to what is considered by the user an optimal setting for aparticular acoustic environment, the user can operate the switch 35,leading to storage of that setting and optionally data indicative of theparticular acoustic environment in the data memory means 34. Actuationof the switch 35 is taken by the sound processor unit 33 as indicatingthat the particular setting of the control means 36 at that time isconsidered optimal by the user for the particular acoustic environmentin which the user finds themself.

The control means 36 can comprise a rotary wheel control that may bemounted on the housing of the prosthesis 30. The switch 35 can also bemounted on the housing of the prosthesis 30. While not depicted, theswitch and controller could be mounted on a remote unit.

The settings selected by the user as being optimal to that user for aplurality of acoustic environments are stored in the data memory means34 optionally with data indicative of those particular acousticenvironments. The sound processor 33 continuously operates in the secondmode, and can simultaneously be operable in the first mode for a definedperiod of time or can be considered as operating in that mode every timethe user adjusts the control means 36 and selects what is considered anew optimal setting for that particular environment. The training periodcan be as long as the user wishes or optionally, the prosthesis 30 cansignal to the user when the training period is complete. In this case,the prosthesis 30 can be adapted to indicate the training period iscomplete once there has been a particular pre-defined number ofinstances in which the user has been unable to select a setting that issubstantially different from the setting already estimated by theprosthesis to be optimal.

The prosthesis 30 further comprises a sound analysis module 37. Thesound analysis module 37 receives input signals from the microphone 27and monitors the acoustic environment of the prosthesis user. The soundanalysis module provides output(s) representative of the type of theacoustic environment being monitored at that time. In the depictedembodiment, the data memory means 34 comprises five data memorylocations. In this embodiment, the first data memory means 34 a containsthe audiometric data of the prosthesis user and/or individual data forone or more loudness models used by the prosthesis 30. The second datamemory location 34 b contains characteristic data about the hearingprosthesis 30. The third data memory location 34 c comprises one or moreequations used to predict the optimal processing operation of the soundprocessor unit 33 for an individual user in different acousticenvironments. The fourth data memory location 34 d stores the optimalsound processing data as selected by the user. The acoustic environmentdata that corresponds to the optimal sound processing data stored in thefourth data memory location is optionally stored in the fifth datamemory location 34 e.

In the depicted embodiment, the fourth data memory location 34 d storesa predefined maximum number of 400 sets of optimal sound processingdata. Other maximum numbers can, however, be envisaged. The depictedembodiment uses a data processing unit 38, that is described in moredetail below. The unit 38 does not utilise all stored data sets but onlya predefined number of most recently logged data sets. In this case, thedata processing unit 38 only utilises the last 256 data sets whencalculating the optimal equation coefficients. Once the memory locationis full, new data sets are stored in the memory location by overwritingthe oldest data set in the memory location. This first in first outstorage system ensures only the most recently logged data is ever storedin the prosthesis 30 at any one time. In another embodiment, older datacannot be overwritten such that once the memory location is full nofurther data sets can be logged by the prosthesis. In anotherembodiment, rules are used to determine the old data that is overwrittenby the newest data set, based on the worth of each old data setaccording to predetermined criteria.

As indicated above, the prosthesis further comprises a data processingunit 38. The data processing unit 38 receives the output of the soundanalysis module 37. Based on the output of the sound analysis module 37,the data processing unit 38 may be adapted to calculate the loudness ofsounds present at the microphone 27. In the depicted embodiment, thedata processing unit 38 may calculate the loudness of the sounds as theywould appear to a person with normal hearing and/or to a person with adegree of hearing impairment. The data processing unit 38 can be adaptedto calculate other acoustic and psychoacoustic measures of the soundspresent at the microphone 27. The data processing unit 38 can use themeasures of the acoustic conditions as inputs to the one or moreequations stored in the third data memory location 34 c which estimatethe optimal sound processing operation of the sound processor 33 for theuser in the acoustic environments determined by the sound analysismodule 37.

The data processing unit 38 further uses the hearing prosthesischaracteristic data stored in the second data memory location 34 b andthe optimal sound processor data generated by the equations toautomatically and continuously determine the appropriate setting of thesound processor 33 to provide the optimal output signal for the user inthe current acoustic environment being experienced by the user.

As described, the sound processor 33 includes an amplifier means and acontrol means 36. At all times the data processing unit 38 operates inthe second mode, where it automatically calculates the operation of thesound processor 33 that is most likely to provide the optimalamplification characteristics in any environment, as represented by theoutput of sound analysis unit 37. The prosthesis can also simultaneouslyoperate in the first mode, where operation of the amplifier means isalso adjustable by the user varying control 36 to allow the user tooptimise the amplification characteristics in the current acousticenvironment.

Where the prosthesis 30 is used in any environment, including anenvironment never previously experienced by the user, the desired gainof the amplifier is calculated through use of an equation having apre-defined form. In this embodiment, the equation is:

G _(i) =a _(i) +b _(i)*max(L _(i) ,c _(i))+d_(i)*(SNR_(i)−SNR_(av))  (1)

where

-   -   i=the band number;    -   G_(i)=gain for band i;    -   a_(i)=trainable coefficient for band i;    -   b_(i)=trainable coefficient for band i;    -   c_(i)=trainable coefficient for band i;    -   d_(i)=trainable coefficient for band i;    -   L_(i)=sound pressure level at the microphone in band i;    -   SNR_(i)=signal to noise ratio in band i; and    -   SNR_(av)=average SNR in all bands.

For each vote, the following training data is stored for each band i:

the sound pressure level or SPL, L_(i), in dB SPL;

the signal-to-noise ratio in band i minus the average for all bands,that is SNR_(i)−SNR_(av), in dB; and

the preferred gain G_(i) also in dB.

Each of the above values are calculated based on the signal at themicrophone 27 for a period of time that precedes the actual vote. Thevalues may be calculated based on the signal at the microphone 27 for atime preceding the vote and/or for a time after the vote has been cast.The stored Li is the average or RMS value of the SPL over thepredetermined period of time and the stored Gi is the average value ofthe gain over a predetermined period of time prior to the actuation ofthe voting button plus the user's preferred gain adjustment at the timeof actuation of the voting button. The SNR in band i is currentlycalculated at the SPL at the microphone exceeded 10% of the time in bandi minus the SPL at the microphone exceeded 90% of the time in band iover the predetermined period of time prior to the vote. This differencemay be more accurately described as an estimate of the modulation depthof the signal. It is envisaged that a more accurate SNR or evenspeech-to-noise ratio estimation techniques may be employed.

The stored training data from many votes, for example up to 256, is usedto form a set of simultaneous equations (1). The set of simultaneousequations is stored as a set of linear equations, however sincecoefficient b_(i) and coefficient c_(i) are potentially multiplied byeach other, that is not linear, the simultaneous equations are initiallylinearised. In other words the value of c_(i) is set to a predefinedvalue, and the MAX [L_(i), c_(i)] operation performed with this value ofc_(I) and the stored L_(i) values in order to set the bracketed term, inother words the greater of L_(i) and c_(i) the result of which isL′_(i). Subsequently the optimal values of a_(i), b_(i) and d_(i) arecalculated with a direct numerical method wherein the optimalcoefficient values are the ones that give the least squared errorbetween the stored G_(i) values and G_(i) values that are calculatedwith the corresponding L′_(i) and stored SNR_(i)−SNR_(av) values. Thisis repeated for at least four alternative values of c_(i), for example35, 45, 55 and 65 dB SPL, and the set of coefficients that gives theleast squared error is selected as the optimal set and applied to theprocessing of the signal, unless the value of b_(i) is outside the rangeof −1.0 to +1.0, or if less than fifty votes have occurred. Otherversions may apply the trained coefficients after more or less thanfifty votes.

The direct numerical method used is lower-upper (LU) decomposition andback substitution which is a variant of Gaussian elimination the set ofsimultaneous equations are stored as matrices in memory, Ax=b, wherematrix A contains the stored acoustic environment data, x is a vectorthat contains the coefficients to be calculated (that is a_(i), b_(i)and d_(i)), and b is a vector that contains the stored G_(i) values. TheLU decomposition method operates on these matrices and solves vector x.Other variants of Gaussian elimination or other direct numerical methodscan be used in other applications.

As an example, the training data from four votes may result in the setof simultaneous equations as follows:

25=a _(i)+(b _(i)*35)+(d _(i)*0)

20=a _(i)+(b _(i)*45)+(d _(i)*0)

15=a _(i)+(b _(i)*55)+(d _(i)*0)

10=a _(i)+(b _(i)*65)+(d _(i)*0)

In this case, the trained coefficients would be a_(i)=42.5, b_(i)=−0.5,c_(i)=35 and d_(i)=0.0.

This set of coefficients would give 0 dB error. In practical situations,there may be more variability in people's amplification preferences, orscatter in the training data. Thus the preferred, user adjusted, G_(i)values on the left of the equal signs in the above equations would notgo up so uniformly in even steps, and the trained coefficient valueswould be different.

Other forms of the equation with other acoustic or psychoacousticparameters, such as higher-order moments of the spectrum of the signaland variations of these moments with time, or other statisticalparameters or estimates of the acoustic signal or combinations of theseparameters, and optionally with other additional coefficients, cancalculate the gain or processing factors other than gain, such as thespeed at which the prosthesis reacts to change in the acousticenvironment, the choice of the equations used will depend upon theparticular application desired, with such applications falling withinthe scope of the present invention.

In the above embodiment, trainable coefficients include a, b, c and d,which as a result leads to the amplifier gain G being a variableprocessing factor.

More than one processing factor of the sound processor 33 can beautomatically calculated using equations similar to equation 1, and thenfurther adjusted by the user using the control means 36 or multiplecontrol means. In this embodiment, the control means 36 can allow theuser to adjust one or more of the following operations of the soundprocessor 33:

(i) the volume of the output signal;

(ii) the gain of the output signal at particular frequencies relative toother frequencies, for example the mid frequency gain can be boosted orattenuated with respect to the low or high band frequencies of theoutput signal; and

(iii) a slope control where the low and high frequency band gains areadjusted in opposing directions while the mid band gain is unchanged.

The user can select which setting of the control means 36 is the optimalone for the particular acoustic environment that they are in byactuating the switch 35. Each time the switch 35 is actuated, the gainin each frequency band is logged along with a data set indicative of theacoustic environment detected by the microphone 27. This data can beused with previous sets of data by the data processing unit 38 tocalculate the gain equation coefficients that enable the equations tobest predict the preferences of the user from the logged training datain each band. Because the data processing unit 38 has access to thepreferred amplification characteristics and the acoustic environments inwhich they were preferred, the data processing unit 38 can calculategeneral relationships between the amplification characteristics and themeasured aspects of the acoustic environment. The automaticallycalculated amplification characteristics can thus vary in a complex butdefined manner as the acoustic environment changes. For example, in thecurrent embodiment, gain at each frequency varies with the overall inputlevel and the signal-to-noise ratio at said frequency relative to thesignal-to-noise ratio at other frequencies.

The depicted data processing unit 38 does not calculate optimal gainequation coefficients until a predetermined number of selections havebeen made by the user. In the depicted embodiment, the prosthesis 30 isadapted to not calculate optimal gain equation coefficients until atleast fifty selections have been made by the user. In this case, wherethe predetermined number of selections have not been made, the soundprocessor 33 will preferably output a signal calculated on the basis ofthe initial, pre-defined values of trainable coefficients. Theseinitial, pre-defined values are calculated for each user by conventionalmethods of prescribing prosthesis operation, or by an empirical, trialand error adjustment process. Once the predetermined number ofselections have been made, the data processing unit 38 re-calculates thetrainable coefficients immediately after every occasion on which theuser operates the switch 35 to indicate that the control 36 is in theoptimal position. In another embodiment, the data processing unit 38does not wait until a predetermined number of selections have been madeby the user to re-calculate the trainable coefficients, but does notapply these coefficients until the equations can accurately predict thepreferences of the aid user with the trained coefficients, and/or theuser has made selections in a preferred minimum range of acousticenvironments.

In an alternative embodiment, a recursive-averaging equation is used todetermine optimal (or trained) values for one or more coefficients.

For example, if the coefficient is the gain below the compressionthreshold (CT), then the optimal, trained setting can be determinedusing the equation:

g(n)=g(n−1)+w*y*ΔG  (2)

where

-   -   n is the vote number    -   g(n) is the trained gain below the CT after vote n    -   w is a weighting factor    -   ΔG is the adjustment made by the user to the variable processing        factor, gain.    -   y=[90−L]/[90−CT] (calculated y set to within bounds: 0.0≦y≦1.0)

Similarly, if the coefficient is the compression ratio (CR), then theoptimal, trained setting can be determined using the equation:

1/CR(n)=1/CR(n−1)+[w*ΔG*(1−2y)]/[90−CT]  (3)

where CR(n) is the trained CR after vote n

Similarly, if the coefficient is the compression ratio (CR), then theoptimal, trained setting can be determined using the equation:

1/CR(n)=1/CR(n−1)+[w*ΔG*(1−2y)]/[90−CT]  (3)

where CR(n) is the trained CR after vote n

The trained values of the coefficients, as calculated with therecursive-averaging equations above, are then converted intocoefficients of the following gain equation:

G=a+b*MAX[L,c]  (4)

In this example, c=CT, b=[1/CR(n)]−1, and a=g(n)−[b*c].

The aid user adjusts the output of the aid to his/her preference (ineffect the aid user is adjusting the applied gain). After voting, theaid uses this output adjustment to calculate the trained compressionratio and gain below the compression threshold with the aboverecursive-averaging equations.

Therefore, a single output (or gain) adjustment by the aid user has beenused to train two coefficients—the CR(n) and the gain below the CT,g(n). The CR(n) and g(n) values are used by the aid to continuously andautomatically calculate the gain that is applied by the aid, G, with Eq.4.

Note that Eq. 4 is very similar to the gain equation that is trained inthe simultaneous equations method (a=g(n)−[b*c], b=[1/CR(n)]−1, andc=CT). Therefore, Eqs. 2 and 3 replace the simultaneous equations usedin the earlier embodiment, and do not require the storage of a finitenumber of sets of preferred gain and corresponding acoustic environmentdata in the aid's memory.

The above can be repeated for i frequency channels. For example, theremay be three frequency channels (low, mid, high) and the aid user mayadjust the aid's output in each of these channels. The output adjustmentfor each channel would then be used to calculate the trained values ofCR(n)_(i) and g(n)_(i) for each channel, and then a_(i) and b_(i) foreach channel. The aid would then automatically calculate the appliedgain, Gi, for each of the i channels many times per second for thecurrent value of the input level, Li.

Referring to FIG. 6, the dot shows the preferred output level that wasvoted for by the aid user for the input level, L, that existed at thetime. The solid line shows the input-output function before the vote,and ΔG is the gain adjustment that was voted for by the aid user. Thedashed line shows the newly trained input-output function that may existafter this vote using the above recursive-averaging equations.

It will be appreciated that direct user adjustments can be made of theaid's settings (to his/her preference) and used to calculate an optimalset of coefficients with some sort of algorithm. The training algorithmscan also be used to calculate the coefficients of other (i.e. non-gain)equations, as well as the values of variable processing factors that arenot calculated with an equation, such as a global volume adjustment orthe T level for an electrode of a cochlear implant.

In an alternative embodiment, the data processing unit 38 does not usethe equations stored in the third data memory location 34 c to determinethe optimal variable processing factors for different environments. Theoptimally adjusted variable processing factors are stored in the fourthdata memory location 34 d at locations determined by the correspondingacoustic environment data supplied by the sound analysis means 37. Priorto the commencement of operation in the first mode, the fourth datamemory locations 34 d are loaded with processing factors that arederived from empirical observations or prescriptive procedures for thecorresponding acoustic environment parameter values. When operating inthe first mode the optimally adjusted processing factor(s) are stored inthe fourth data memory 34 d at locations determined by the acousticenvironment data supplied by the sound analysis means 37. For example,the fourth data memory location 34 d can store a multi-dimensional (orN-dimensional) look-up table or matrix, where each dimension is adifferent acoustic environment or psychoacoustic parameter, and theparameter values determine the location in the table, matrix, or memorylocation 34 d where the preferred processing factor(s) is stored. Thusthe matrix is indexed by rounded values of the N acoustic environmentparameters (or optionally the estimated psychoacoustic parameters suchas loudness). The optimally adjusted processing data may simplyoverwrite the prescribed data in the fourth data memory 34 d, or may becombined with old data written to the same location using a mathematicaland/or statistical procedure. When operating in the second mode, theacoustic environment and psychoacoustic parameter values supplied by thesound analysis means 37 are used to index the table or matrix or fourthdata memory location 34 d, and the value(s) stored at the indexedlocation are used to set the target value(s) of the correspondingprocessing factor(s) for the sound processing means 33. The processingfactor(s) applied in the sound processing means 33 may slowly convergetowards the target value(s) to avoid the possibility of undesirableacoustic effects associated with excessively rapid or instantaneouschanges of the applied value(s). For memory locations that contain aprescribed value of a processing factor, the target value can beadjusted to represent the trend of user-adjusted values for environmentslocated near the current environment in the memory location 34 d. Forthis alternative embodiment there is no need for the fifth data memorylocation 34 e and the equations stored in the third data memory 34 c,and the variation of a processing factor with acoustic parameterssupplied by the sound analysis means 37, or psychoacoustic parameterscalculated by the data processing unit 38, is not restricted by therelationships that are predefined in the equations stored in the thirddata memory location 34 c.

In one embodiment, the hearing prosthesis can be adapted to output audiosignals to a single ear of the user. In another embodiment, the hearingprosthesis can be adapted to output audio signals to both ears of theuser.

In a further embodiment, the trainable hearing prosthesis can be adaptedto accept microphone signals from microphones mounted on each side ofthe head, and then to output signal to either or both ears.

FIG. 3 is a flowchart of the logic behind the operation of the dataprocessing unit 38.

The Start of the flowchart is when the prosthesis 30 is turned on andpower is supplied to the parts thereof including the sound processor 33and data processing unit 38. The end point of the flowchart is when theuser turns the prosthesis 30 off or if the battery for the power sourcegoes flat. While not depicted, it is possible that another end pointcould be defined as when a certain number of votes are reached, or ifthe user stops training the prosthesis 30 for a predetermined period oftime, or if the prosthesis 30 determines from the user adjustments ofthe control means 36 that it is sufficiently well-trained. Stillfurther, an audiologist or other third party could disable furthertraining of the prosthesis when the user is satisfied with theperformance of the prosthesis 30 in a range of different acousticenvironments. However, it is desirable that the user can train theprosthesis at any time, so that training does not need to be re-enabledif the user encounters a new environment or if the hearing loss profileof the user changes over time or with slow acclimatisation to theprosthesis.

FIG. 4 is a graph of the results of preliminary experiments performed ona prototype prosthesis according to the present invention. In theexperiments, the goal was to train the prosthesis to produce a greatergain than provided by the prosthesis prior to the training in allfrequency bands for any input sound. FIG. 4 indicates how the necessityto adjust the control 36 (RMS control adjustment in dB) following groupsof 15 votes to get the desired output decreased due to the “training”ability of the prosthesis.

FIG. 5 is an illustration of the matrix previously referred to usingacoustic parameters of equation 1.

FIG. 5 is a diagram representing the storage of processing factor valuesin a two-dimensional memory space that is indexed by the values of twoacoustic parameters, the sound pressure level for band i and thesignal-to-noise ratio for band i relative to the average for all bands,although this does not exclude the use of a greater number of dimensionsfrom the scope of this invention, and the values shown are illustrativeonly. The numbers in the memory locations represent gain values, wheregain is a variable processing factor. The stored values are prescribedvalues, except the bold italic values which represent data that differsfrom the initial prescribed data due to the storage of user-preferredsetting data as a result of training by the user.

It will be appreciated by persons skilled in the art that numerousvariations and/or modifications may be made to the invention as shown inthe specific embodiments without departing from the spirit or scope ofthe invention as broadly described. The present embodiments are,therefore, to be considered in all respects as illustrative and notrestrictive.

1-117. (canceled)
 118. An auditory prosthesis for rehabilitating thehearing of a user, comprising: a microphone configured to receive soundand to produce an output signal corresponding to the received sound; asound processor configured to process the microphone signal using a setof variable processing factors, each having a value, to produce anoutput signal; and a sound analyzer configured to receive the microphonesignal, and to output a data set representative of the acousticenvironment of the prosthesis; wherein the prosthesis is configured toprovide the user with the ability to individually adjust the value of atleast one of the variable processing factors for a first acousticenvironment, and wherein the prosthesis is further configured toautomatically make subsequent adjustments to the value of the at leastone variable processing factor for a second acoustic environment basedon adjustments of the value of the at least one variable processingfactor by the user and the acoustic environment in which the value ofthe at least one variable processing factor was adjusted.
 119. Theprosthesis according to claim 118, further comprising: a data memoryconfigured to store said data set output by the sound analyzer and theadjusted value of at least one variable processing factor.
 120. Theprosthesis according to claim 118, wherein the prosthesis is furtherconfigured to output audio signals to a single ear of the user.
 121. Theprosthesis according to claim 118, wherein the prosthesis is furtherconfigured to output audio signals to both ears of the user.
 122. Theprosthesis according to claim 119, wherein the prosthesis is configuredprovide the user with the ability to repeatedly adjust the value of theat least one variable processing factor and to monitor how thepreference of the user for a particular setting of the at least onevariable processing factor changes with time or usage.
 123. Theprosthesis according to claim 122, wherein the number of adjustmentsmade by the user is monitored by the prosthesis.
 124. The prosthesisaccording to claim 123, further comprising a control means configured toenable the user to adjust the at least one variable processing factor.125. The prosthesis according to claim 118, wherein the auditoryprosthesis is a hearing aid.
 126. The prosthesis according to claim 125,further comprising an output device configured to provide the outputsignals to the user.
 127. The prosthesis according to claim 126 whereinthe output device is an earphone that receives the output signal of thesound processing means and generates amplified sounds that are deliveredinto the ear of the user.
 128. The prosthesis according to claim 126,wherein the output device is a vibrating mechanism mechanically coupledto the middle or inner ear.
 129. The prosthesis according to claim 126,wherein the output device comprises a receiver/stimulator unit thatreceives encoded stimulation data from the sound processing means andoutputs stimulation signals that are delivered to a cochlea of the userthrough an electrode array.
 130. The prosthesis according to claim 129,wherein the sound processor is configured to use a coding strategy toextract speech from the sounds detected by the microphone before orafter processing of the microphone signal according to the set ofvariable processing factors.
 131. The prosthesis according to claim 130,wherein the speech processor is configured to perform an audio spectralanalysis of the microphone signal to generate output channel amplitudelevels.
 132. The prosthesis according to claim 131, wherein thetransformation from the microphone signal to output channel amplitudelevels is based on the set of variable processing factors.
 133. Theprosthesis according to claim 132, wherein the channel amplitude levelsinclude a threshold level and discomfort level that are based onprevious adjustments of the at least one variable processing factor madeby the user.
 134. The prosthesis according to claim 129, wherein thereceiver/stimulator unit is positioned within a housing that isimplantable within the user.
 135. The prosthesis according to claim 134,wherein the housing for the receiver/stimulator unit is implantablewithin a recess in the bone behind the ear posterior to the mastoid.136. The prosthesis according to claim 131, wherein the speech processoris configured to encode the sound detected by the microphone into asequence of electrical stimuli in accordance with a predeterminedalgorithm.
 137. The prosthesis according to claim 136, furthercomprising: a transmitter coil; and wherein the implantedreceiver/stimulator unit comprises a receiver coil; and whereinprosthesis is configured to transfer the encoded sequence to theimplanted receiver/stimulator unit using the transmitter coil and thereceiver coil of the receiver/stimulator unit.
 138. The prosthesisaccording to claim 118, wherein the auditory prosthesis is a cochlearimplant.
 139. A method of rehabiliating the hearing of a user,comprising: converting a received sound to a microphone signal;processing the microphone signal at a speech processor to generate anoutput signal utilizing a set of variable processing factors, receivingfrom a user an adjustment of the individual value of at least one of thevariable processing factors of the set; and determining a data setrepresentative of the acoustic environment of the prosthesis in whichthe adjustment was received from the user based on the microphonesignal; automatically adjusting one or more of the variable processingfactors for a second acoustic environment based on the receivedadjustments of the at least one variable by the recipient and theacoustic environment in which the value of the at least one variableprocessing factor was adjusted